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• 12.10 Quantal display issues using nvidiaXineramaInfoOverride

  - by AvatarKava

  After updating to 12.10 today, my xorg.conf doesn't seem to be respected by Quantal. Not sure if this is a 'bug' or whether it's just an adjustment I have to make due to changes in the OS. When logging in, it seems Ubuntu is now recognizing only one 3840x1080 screen named "Matrox" and maximizing windows spans them across both screens. In 12.04, this configuration file successfully allowed me to override the data provided by my TripleHead2Go and maximize windows to a single monitor. Any ideas or where to start on trying to debug this? After a bit of searching, I tried to make changes according to the update here: http://www.phoronix.com/scan.php?page=news_item&px=MTEyMDk Here's where the config file sits currently: Section "ServerLayout" Identifier "Layout0" Screen 0 "Screen0" 0 0 InputDevice "Keyboard0" "CoreKeyboard" InputDevice "Mouse0" "CorePointer" Option "Xinerama" "0" EndSection Section "Files" EndSection Section "InputDevice" Identifier "Mouse0" Driver "mouse" Option "Protocol" "auto" Option "Device" "/dev/psaux" Option "Emulate3Buttons" "no" Option "ZAxisMapping" "4 5" EndSection Section "InputDevice" Identifier "Keyboard0" Driver "kbd" EndSection Section "Monitor" # HorizSync source: edid, VertRefresh source: edid Identifier "Monitor0" VendorName "Unknown" ModelName "Matrox" HorizSync 31.5 - 80.0 VertRefresh 59.9 - 75.0 Option "DPMS" EndSection Section "Device" Identifier "Device0" Driver "nvidia" VendorName "NVIDIA Corporation" BoardName "GeForce GTX 260M" EndSection Section "Screen" Identifier "Screen0" Device "Device0" Monitor "Monitor0" DefaultDepth 24 Option "nvidiaXineramaInfo" "true" Option "nvidiaXineramaInfoOrder" "CRT-0" #Option "metamodes" "CRT: nvidia-auto-select +0+0" Option "nvidiaXineramaInfoOverride" "1920x1080 +0+0, 1920x1080 +1920+0" Option "Stereo" "0" SubSection "Display" Depth 24 EndSubSection EndSection

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• Why does ffmpeg stop randomly in the middle of a process?

  - by acidzombie24

  ffmpeg feels like its taking a long time. I then look at my output file and i see it stops between 6 and 8mbs. A fully encoded file is about 14mb. Why does ffmpeg stop? My code locks up on StandardOutput.ReadToEnd();. I had to kill the process (after seeing it not move for more then 10 seconds when i see it update every second previously) then i get the results of stdout and err. stdout is "" stderr is below. The output msg shows the filesize ended. I also see a drop in my CPU usage when it stops. I copyed the argument from visual studios. CD to the same working directory and ran the cmd (bin/ffmpeg) and pasted the argument. It was able to complete. int soundProcess(string infn, string outfn) { string aa, aa2; aa = aa2 = "DEAD"; var app = new Process(); app.StartInfo.UseShellExecute = false; app.StartInfo.RedirectStandardOutput = true; app.StartInfo.RedirectStandardError = true; //*/ app.StartInfo.FileName = @"bin\ffmpeg.exe"; app.StartInfo.Arguments = string.Format(@"-i ""{0}"" -ab 192k -y {2} ""{1}""", infn, outfn, param); app.Start(); try { app.PriorityClass = ProcessPriorityClass.BelowNormal; } catch (Exception ex) { if (!Regex.IsMatch(ex.Message, @"Cannot process request because the process .*has exited")) throw ex; } aa = app.StandardOutput.ReadToEnd(); aa2 = app.StandardError.ReadToEnd(); app.WaitForExit(); if (aa2.IndexOf("could not find codec parameters") != -1) return 1; else if (aa == "DEAD" || aa2 == "DEAD") return -1; else if (aa2.Length != 0) return -2; else return 0; } The output of stderr. stdout is empty. FFmpeg version SVN-r15815, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-memalign-hack --enable-postproc --enable-swscale --enable-gpl --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libx264 --enable-libxvid --disable-ffserver --disable-vhook --enable-avisynth --enable-pthreads libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 6. 1 / 0. 6. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 13 2008 10:28:29, gcc: 4.2.4 (TDM-1 for MinGW) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\dev\src\trunk\prjname\prjname\App_Data/temp/m/o/6304266424778814852': Duration: 00:12:53.36, start: 0.000000, bitrate: 154 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Output #0, ipod, to 'C:\dev\src\trunk\prjname\prjname\App_Data\temp\m\o\2.m4a': Stream #0.0(und): Audio: libfaac, 44100 Hz, stereo, s16, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 87kB time=4.74 bitrate= 150.7kbits/s size= 168kB time=9.06 bitrate= 151.9kbits/s size= 265kB time=14.28 bitrate= 151.8kbits/s size= 377kB time=20.29 bitrate= 152.1kbits/s size= 487kB time=26.22 bitrate= 152.1kbits/s size= 594kB time=32.02 bitrate= 152.1kbits/s size= 699kB time=37.64 bitrate= 152.1kbits/s size= 808kB time=43.54 bitrate= 152.0kbits/s size= 930kB time=50.09 bitrate= 152.2kbits/s size= 1058kB time=57.05 bitrate= 152.0kbits/s size= 1193kB time=64.23 bitrate= 152.1kbits/s size= 1329kB time=71.63 bitrate= 152.0kbits/s size= 1450kB time=78.16 bitrate= 152.0kbits/s size= 1578kB time=85.05 bitrate= 152.0kbits/s size= 1706kB time=92.00 bitrate= 152.0kbits/s size= 1836kB time=98.94 bitrate= 152.0kbits/s size= 1971kB time=106.25 bitrate= 151.9kbits/s size= 2107kB time=113.57 bitrate= 152.0kbits/s size= 2214kB time=119.33 bitrate= 152.0kbits/s size= 2345kB time=126.39 bitrate= 152.0kbits/s size= 2479kB time=133.56 bitrate= 152.0kbits/s size= 2611kB time=140.76 bitrate= 152.0kbits/s size= 2745kB time=147.91 bitrate= 152.1kbits/s size= 2880kB time=155.20 bitrate= 152.0kbits/s size= 3013kB time=162.40 bitrate= 152.0kbits/s size= 3146kB time=169.58 bitrate= 152.0kbits/s size= 3277kB time=176.61 bitrate= 152.0kbits/s size= 3412kB time=183.90 bitrate= 152.0kbits/s size= 3540kB time=190.80 bitrate= 152.0kbits/s size= 3670kB time=197.81 bitrate= 152.0kbits/s size= 3805kB time=205.08 bitrate= 152.0kbits/s size= 3932kB time=211.93 bitrate= 152.0kbits/s size= 4052kB time=218.38 bitrate= 152.0kbits/s size= 4171kB time=224.82 bitrate= 152.0kbits/s size= 4277kB time=230.55 bitrate= 152.0kbits/s size= 4378kB time=235.96 bitrate= 152.0kbits/s size= 4486kB time=241.79 bitrate= 152.0kbits/s size= 4592kB time=247.50 bitrate= 152.0kbits/s size= 4698kB time=253.21 bitrate= 152.0kbits/s size= 4804kB time=258.95 bitrate= 152.0kbits/s size= 4906kB time=264.41 bitrate= 152.0kbits/s size= 5012kB time=270.09 bitrate= 152.0kbits/s size= 5118kB time=275.85 bitrate= 152.0kbits/s size= 5234kB time=282.10 bitrate= 152.0kbits/s size= 5331kB time=287.39 bitrate= 151.9kbits/s size= 5445kB time=293.55 bitrate= 152.0kbits/s size= 5555kB time=299.40 bitrate= 152.0kbits/s size= 5665kB time=305.37 bitrate= 152.0kbits/s size= 5766kB time=310.80 bitrate= 152.0kbits/s size= 5876kB time=316.70 bitrate= 152.0kbits/s size= 5984kB time=322.50 bitrate= 152.0kbits/s size= 6094kB time=328.49 bitrate= 152.0kbits/s size= 6212kB time=334.76 bitrate= 152.0kbits/s size= 6327kB time=340.99 bitrate= 152.0kbits/s

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• Programmatically convert WAV

  - by kaykun

  Hi, I'm writing a file compressor utility in C++ that I want support for PCM WAV files, however I want to keep it in PCM encoding and just convert it to a lower sample rate and change it from stereo to mono if applicable to yield a lower file size. I understand the WAV file header, however I have no experience or knowledge of the actual sound data works. So my question is, would it be relatively easy to programmatically manipulate the "data" sub-chunk in a WAV file to convert it to another sample rate and change the channel number, or would I be much better off using an existing library for it? If it is, then how would it be done? Thanks in advance.

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• iPhone metering problems

  - by Eric Christensen

  When I'm recording an AVAudioRecorder object, I repeat a chunk of code via an NSTimer that includes: [soundrecording updateMeters]; NSLog(@"channel 0 average:%f, peak:%f",[soundrecording averagePowerForChannel:0],[soundrecording peakPowerForChannel:0]); NSLog(@"channel 1 average:%f, peak:%f",[soundrecording averagePowerForChannel:1],[soundrecording peakPowerForChannel:1]); When I'm recording a mono file, the peak power for channel 0 is just what you'd expect, a float from -160 to 0. But average power for channel 0 is always zero. (And, of course, the values for channel 1 are both zero.) When I'm recording a stereo file, both the average and peak values for both channels are as expected. Any thoughts on why, when recording a mono file, the average value for channel 0 isn't returning correctly, even though the peak is? Thanks!

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• Play wave file using AudioFormat in java

  - by angelina

  Dear all, I m getting following exception while running my code on linux operating system.This code works fine on windows operating system.below is the exception and code used. java.lang.IllegalArgumentException: No line matching interface Clip supporting format PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian is supported. AudioFormat format = sourceaudio.getFormat(); format = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED, format.getSampleRate(), format.getSampleSizeInBits() * 2, format.getChannels(), format.getFrameSize() * 2, format.getFrameRate(), true); AudioFileFormat.Type targettype = AudioFileFormat.Type.WAVE; AudioInputStream targetaudiostream = AudioSystem.getAudioInputStream(format, sourceaudio); sourceaudio.close(); targetaudiostream.close(); System.out.println("55555555"); URL url = new URL("http://localhost:8084/newvideo/PCMfile.wav"); Clip clip = AudioSystem.getClip(); AudioInputStream ais = AudioSystem.getAudioInputStream(url); clip.open(ais); System.out.println("seconds: " + (clip.getMicrosecondLength() / 1000000));

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• Resampling audio output for A2DP (from PCM WAV)

  - by user1669982

  The question is how to bring stereo PCM WAV 32,000 Hz with a stream of 1024 kbps (125 KB) to the headset with Bluetooth 2.1 on a CM7 smartphone with DSPManager. Is it possible? SBC is really bad idea. To TJD: Because it compresses the compressed stream. My Epic 4G don`t have Apt-X support. My headset Gemix BH-04A yellow. May be its possible with the Headset Profile (HSP)? I dont know about supported codecs in this profile.

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• The best Bar on the globe is ... in Seoul/Korea

  - by Mike Dietrich

  As you know already sometimes I write about things which really don't have to do anything with a database upgrade. So if you are looking for tips and tricks and articles about that topic please stop reading now Actually I'm not a lets-go-to-a-bar person. I enjoy good food and a fine dessert wine afterwards. But last week in Seoul/Korea Ryan, our local host, did ask us after a wonderful dinner at a Korean Barbecue place if we'd like to visit a bar. I was really tired as I flew into Seoul overnight from Sunday to Monday arriving Monday early morning, getting shower, breakfast - and then a full day of very good and productive customer meetings. But one thing Ryan mentioned catched my immediate attention: The owner of the bar collects records and has a huge tube amp stereo system - and you can ask him to play your favorite songs. The bar is called "Peter, Paul and Mary" - honestly not my favorite style of music. And I even coulnd't find a webpage or an address - only that little piece of information on Facebook. But after stepping down the stairs to the cellar my eyes almost poped out of my head. This is the audio system: Enourmus huge corner horn loudspeakers from Western Electric. Pretty old I'd suppose but delivering an incredible present dynamics into the room. And plenty of tube equipment from Jadis, NSA Labs and Shindo Laboratories Western Electric 300B Limited amps from Tokyo. And the owner (I was so amazed I had simply forgotten to ask for his name) collects records since 40 years. And we had many wishes that night. Actually when we did enter Peter, Paul and Mary he played an old Helloween song. That must have been destiny. A German entering a bar in Korea and the owner is playing an old song by one of Germany's best heavy metal bands ever. And it went on with the Doors, Rainbow's Stargazer, Scorpions, later Deep Purple's Perfect Strangers, a bit of Santana, Carly Simon, Jimi Hendrix, David Bowie ...Ronnie James Dio's Holy Diver, Gary Moore, Peter Gabriel's San Jacinto ... and many many more great songs ... Of course we were the last guests leaving the place at 2am in the morning - and I've never ever had a better night in a bar before ... I could have stayed days listening to so many records  ... Thanks Ryan, that was a phantastic night! -Mike

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• How can I fix these errors with Panda3D's sample projects?

  - by lhk

  I just installed the latest Panda3D packages on a Mint 12 32-bit virtual machine. Then I downloaded and configured Eclipse and tried to run the Asteroids sample project. The window is created properly. But after rendering the scence once the game freezes. This happens with the other sample apps, too. Here's the error log: DirectStart: Starting the game. Known pipe types: glxGraphicsPipe (all display modules loaded.) :display:gsg:glgsg(warning): Occlusion queries advertised as supported by OpenGL runtime, but could not get pointers to extension functions. OpenGL Warning: glXChooseFBConfig returning NULL, due to attrib=0x6, next=0xffffffff :display:glxdisplay(warning): No suitable FBConfig contexts available; using XVisual only. depth_bits=16 color_bits=24 alpha_bits=8 stencil_bits=8 accum_bits=64 back_buffers=1 stereo=1 force_hardware=1 AL lib: pulseaudio.c:331: PulseAudio returned minreq > tlength/2; expect break up :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant :display:gsg:glgsg(error): at 5703 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4654 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4654 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant :display:gsg:glgsg(error): at 5703 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 3057 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 3057 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant OpenGL Warning: No pincher, please call crStateSetCurrentPointers() in your SPU :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant :display:gsg:glgsg(error): at 5703 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 5703 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 3661 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 3661 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant :display:gsg:glgsg(error): at 4765 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid enumerant :display:gsg:glgsg(error): at 5703 of panda/src/glstuff/glGraphicsStateGuardian_src.cxx : invalid operation :display(error): Deactivating glxGraphicsStateGuardian. What can I do to fix the problem ?

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• How to debug lag using Bluetooth connected mouse and A2DP headset?

  - by gertvdijk

  I own a Logitech M555b mouse (since a week) for use with my HP Elitebook 8570w laptop running Kubuntu 12.04. Works fine right after connecting using the KDE Bluetooth control module. However, after some time (seemingly random), it starts to lag. Movements are being delayed for roughly 500ms for a short period of time. Usually it recovers after some time too, but it can take minutes. All actions are being delayed: movements, click, scrolls. Additionally, the movements can be choppy during these times. A workaround that always works for the same short period of time is to disconnect an re-connect the mouse. This can be done using the same KDE Bluetooth control module. What did I try already? Running this at boot time: echo on > `readlink -f /sys/class/bluetooth/hci0`/../../../power/level To disable any power saving features on the Bluetooth hci0 device. Check the mouse's batteries (it's just a week old, other new batteries: same result) Checking logs and kernel messages about Bluetooth-related entries: none aside the expected messages on connect time. I'm running kernel 3.5.0-13-generic as provided in the xorg-edgers PPA. Booting the regular 3.2 Precise kernel results in the same behaviour. Some other information that may help: It happens when no other Bluetooth connections are active on the machine. Similar symptoms also occur on my Bluetooth stereo (A2DP) headset, but then it's audio lagging and skipping. Swapping Bluetooth profiles as described here then helps. Conclusion: it's not the mouse that's faulty. The headset always worked fine using my now dead Thinkpad T61p with built-in Bluetooth. The bluetooth module in my laptop is connected via USB and shows up as Bus 002 Device 003: ID 0a5c:21e1 Broadcom Corp. I'm mobile and several people around me are using Bluetooth at work (A2DP mostly). It also occurs at home, where my neighbours are probably using Bluetooth as well. It could just be radio interference, but I think Bluetooth connections should just hop to another channel. And, moreover, it just works properly instantly when re-connecting. Therefore I think it's a software driver issue and I'd like to debug it. Is there any way to get more verbose logging on the Bluetooth(-hid) modules?

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• Oracle Service Cloud May 2014 Release – Focus on your driving by JP Saunders

  - by Tuula Fai

  The next time you’re twiddling dials on your car’s dashboard to get the air to blow in the right direction, and the right song to play on the stereo, while pulling on the wires to charge your phone and punching in passwords to re-sync your hands-free headset to take a call, consider this… Does having a better dashboard UI in your car improve your driving performance? The Tesla car has one of the most modern and intuitive dashboards in any commercial car today. It is actually based on the design of a smart phone, which can download apps and updates directly from the cloud.  The 17” touchscreen, Lynx-based dashboard totally integrates all channels and devices, allowing the driver to focus on the smooth driving and power of this luxury (toy) car.  What the folks at Tesla didn't do was avoid the complexity of our needs. Instead, they streamlined them. And, while we might not all be able to afford a Tesla, their approach demonstrates that a modern UI approach can ultimately make a positive difference in our lives and businesses.  This is why the productivity and effectiveness of a Modern Contact Center is many times greater than that of a traditional contact center. Agents in a Modern Contact Center get to focus on the task at hand, the customer engagement, rather than stumbling their way through Lego blocks of complexity.  The Oracle Service Cloud is a modern approach to customer service that empowers your agents to achieve greater focus on improving your operational and strategic success through streamlined business processes.  Here are some of the recent May 2014 release highlights to the Oracle Service Cloud: Performance Enhanced Desktop UI A modern agent desktop interface that optimizes clumsy tasks, logins, screens and workflows and is optimized for agent and system performance. Improvements include performance for drag-and-drop configurable views, saved searches, and improved caching for high-speed performance even during disconnected or slow internet access.  Customer Experience Routing A streamlined automatic way to connect the right customer need to the best agent skills, based on multidimensional variables such as product skills, language skills, workload, call volume to optimize the connection and resolution experience. On-The-Go Mobile Improvements to the Agent mobile app that extend connectivity to websites, and customer surveys that are mobile-ready and rendered for any device, and ensure the customer’s voice is captured while the insight is still top of mind.  Infused Social Engagement Enhancements to infused social capabilities allow agents to respond in social threads directly from within the agent desktop, with the information becoming part of the incident record for automatic actions (such as replay or escalate) triggered off the response. Front-End Siebel Contact Center The market leading online Web Customer Self-Service interface from the Oracle Service Cloud, is now out-of-the-box ready for Oracle Siebel customers. Deploy a new online web self-service interface in a matter of weeks to have customers self-serve and self-solve answers, with escalated incidents routed directly into the Oracle Siebel Contact Center. For more information on the latest enhancements for the Oracle Service Cloud, please see the Oracle Service Cloud May 2014 Capabilities and Benefits. Related blogs: Oracle Service Cloud Feb 2014

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• Misadventures at Radio Shack

  - by Chris Williams

  While I'm waiting for my Arduino kits to show up, I started reading the Getting Started With Arduino book from O'Reilly (review coming later) and I'm about 40 pages in when I get to a parts list for one of the first projects. Looks pretty straightforward, and even has Radio Shack part numbers next to almost everything. So on my lunch today, I decided to run out to "The Shack" (seriously, that's their rebranding?) to pick up some basics, like a couple resistors, a breadboard, a momentary switch and a pack of pre-cut jumper wires. I found the resistors without any difficulty, and while they didn't have the exact switch I wanted, it was easy enough to find one that would do. That's where my good luck abruptly ended. I was surprised that I couldn't find a breadboard or any jumper wires, so while I was looking around, a guy came up and asked me if I needed some help. I told him I did, explained what I needed and even gave him the Radio Shack part number for the pack of jumper wires. After a couple minutes he says he can't find anything in the system, which was unfortunate but not the end of the world.  So then I asked him about the breadboard, and he pointed me to some blank circuitboards (which are not the same thing) and I said (nicely) that those weren't breadboards and attempted to explain (again) what I needed, at which point he says to me "I don't even know what the hell you're looking for!" I stood there for a moment and tried to process his words. About that time, another salesperson came up and asked what I was trying to find. I told her I needed a breadboard, and she pointed to the blank circuit boards and said "they're right in front of you..." After seeing the look on my face, she thought for a minute and said... "OH! you mean those white things. We don't have those anymore." I thanked her, set everything down on the counter and left. (I wasn't going to buy only half the stuff I needed.. and I was pretty sure I was never going to be buying ANYTHING at that particular location ever again.) Guess I'll be ordering more stuff online at this point. It's a shame really, because I used to LOVE going to Radio Shack as a kid, and looking through all the cool electronics components and stuff, even if I didnt understand what most of them were at the time. Seems like the only thing they carry in any quantity/variety now is cell phones and random stereo connectors.

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• Why does my Intel HDA onboard sound card not have a "Mix" device / channel?

  - by Hanno Fietz

  I want to be able to record what my sound card outputs on the speakers / headphones. This question is all over the interwebs again and again, and there seem to be two outcomes: in your selection of audio input devices, there's a device called "Stereo Mix", or similar, which is the "loopback" device for audio. Choose that in your recording tool and you're done. there's no such device and only speculative posts about why that may be. Now, I'm using ALSA and an Intel HDA chipset on my mainboard under Kubuntu Karmic. I have some 5-10 output channels and "Mic", "Front Mic" and "Line" for input. All of those are available in KMix, Audacity and other software. No "loopback" / "Mix" / whatever. Do I have to get some driver / kernel module set up ALSA in some way set up my system configuration in some way use a software solution (such as JACK) I had a look at JACK, and found it rather hard to understand, it's either an expert tool or just clumsy, I couldn't say. At least, I wasn't able to figure out how to achieve what I wanted. One of my problems seems to be that I don't understand where and how the mixing happens. Are there sound cards which just aren't able to do it? Why does the sound card matter at all, since I could in theory grab the data stream at some point before it goes to the hardware, right?

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• How can I stream audio signals from various devices/computers to my home server?

  - by Breakthrough

  I currently have a headless home server set up (running Ubuntu 12.04 server edition) running a simple Apache HTTP server. The server is near an audio receiver, which controls a set of indoor and outdoor speakers in my home. Recently, my father purchased a Bluetooth adapter, which our various laptops and cellphones can connect to, outputting the music to the speakers. I was hoping to find a solution that worked over Wi-Fi, namely because it won't cost anything (I already have a server with an audio card), and it doesn't depend on Bluetooth. Is there any cross-platform (preferably free and open-source) solution that I can use which will allow me to stream audio to my home server, over my home network, from a wide variety of devices (laptops running Windows/Linux or cellphones running Android/BB/iOS)? I need something that works at least with Windows and Android. Also, just to clairfy, I want something that simply allows devices to connect to my server and output an audio signal without any action on the server end (since it's a server hidden away near my receiver). Any subsequent connection attempt should be dropped, so only one device can be in control of the stereo at once.

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• SPDIF passthrough not working in Windows 7

  - by adriangrigore

  Hi, I'm running Windows 7 on a computer with an Audigy Platinum eX sound card connected to a surround receiver via optical cabling. Sound works fine when listening to non-surround audio sources, such as windows sounds or MP3. However, when I view a DVD in Media Center and the SPDIF passthrough kicks in, I can only hear an awful noise instead of the movie soundtrack. Also, the receiver does not show the Dolby Digital or DTS symbol, but stays at Dolby Prologic, so it seems it doesn't identify the sound encoding properly. I could switch off SPDIF passthrough and use the sound card's decoder instead, but that's not an option for me since it would create more problems with regular MP3 playback via additional Stereo Receiver which is also connected to the same sound card. I've tried both the default Audigy drivers that come with Windows 7 and the latest drivers from the Soundblaster website, but the problem remains unchanged. Also, I have ensured that the receiver's Dolby Digital decoder is not broken by successfully connecting it to my PS3 to view a Dolby Digital DVD. Besides, SPDIF passthrough was working fine in Vista before I upgraded to Windows 7. Is there anything else I could try?

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• Convert a cassette tape recording to digital format

  - by Electric Automation

  Has anyone been successful with transferring audio cassette tape recordings to a digital format? I would like to preserve old cassette tape recordings of my grandparents to some digital format: MP3, WAV, etc... The quality of the tapes are mediocre. I think I can handle the quality restoration but getting the audio from tape to digital is my question. Below is a list of the hardware that I can work with: Cassette Deck: I have a Technics stereo cassette deck model RS-B12. It has separate left and right IN and OUT RCA type jacks on the back. In the front it has a headphone phono jack, plus left and right mic input phono jacks. On the computer side: -I have a Windows Vista PC with no additional software other than what came with the machine from Costco. No sound editing software that I can see. There is no sound card on the PC. On the front panel there is a mini-phono mic input jack and there are several different types of in/out mini-phono jacks on the back. In addition, USB and Firewire. I also have access to a new (2009) iMac with a mini-phono input jack for a powered mic or other audio source and GarageBand that has come with the computer. In addition, USB and Firewire. What are my options for getting these cassette recordings into a digital format? Whats the best format? What sort of wires would I need and will I want to utilize the USB or Firewire or can I simply use the audio inputs on the PC (or Mac) to receive the audio stream?

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• AVCHD MTS h264 1080p file with choppy playback in Linux

  - by marc

  When I'm trying play video files from my camera: Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) Input #0, mpegts, from '00027.MTS': Duration: 00:00:38.88, start: 2.884289, bitrate: 16945 kb/s Program 1 Stream #0.0[0x1011]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s … on my Linux computer (Ubuntu 12.04), I get choppy playback. It's completly unusable... I tried: Totem VLC mplayer The result is always same issue. I sent the same video file to a friend who has ubuntu 10.04 to test, and he also has the same issue. He has Windows 7, and confirms that on Windows, the video work well. I have an Intel® Core™2 CPU 6300 @ 1.86GHz × 2 with GF 9600 GT, with closed NVIDIA drivers. This is not any kind of issue with big files playing slow from an HDD issue. I have an SSD drive! I spent the last days and nights, trying hundreds of commands for ffmpeg, handbrake, mencoder... Any of them won't let me create a file with enough quality. I downloaded few movies from YouTube in 1080p, and playback worked well without any big pixels and choppiness. I would like have highest possible quality, I will put following files onto a Blu-ray disk so I don't need to compress them to get a smaller size. I just want smoth playback on my Linux box. On Windows, the same file is working well.

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• Convert a cassette tape recording to digital format

  - by Optimal Solutions

  Has anyone been successful with transferring audio cassette tape recordings to a digital format? I would like to preserve old cassette tape recordings of my grandparents to some digital format: MP3, WAV, etc... The quality of the tapes are mediocre. I think I can handle the quality restoration but getting the audio from tape to digital is my question. Below is a list of the hardware that I can work with: Cassette Deck: I have a Technics stereo cassette deck model RS-B12. It has separate left and right IN and OUT RCA type jacks on the back. In the front it has a headphone phono jack, plus left and right mic input phono jacks. On the computer side: -I have a Windows Vista PC with no additional software other than what came with the machine from Costco. No sound editing software that I can see. There is no sound card on the PC. On the front panel there is a mini-phono mic input jack and there are several different types of in/out mini-phono jacks on the back. In addition, USB and Firewire. I also have access to a new (2009) iMac with a mini-phono input jack for a powered mic or other audio source and GarageBand that has come with the computer. In addition, USB and Firewire. What are my options for getting these cassette recordings into a digital format? Whats the best format? What sort of wires would I need and will I want to utilize the USB or Firewire or can I simply use the audio inputs on the PC (or Mac) to receive the audio stream?

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• How do I convert a video to GIF using ffmpeg, with reasonable quality?

  - by Kamil Hismatullin

  I'm converting .flv movie to .gif file with ffmpeg. ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif It works great, but output gif file has a very law quality. Any ideas how can I improve quality of converted gif? Output of command: $ ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:52:53 with gcc 4.7.2 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.flv': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 Duration: 00:00:18.85, start: 0.000000, bitrate: 3098 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x720, 2905 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 192 kb/s Metadata: creation_time : 2013-02-14 04:00:07 [buffer @ 0x92a8ea0] w:1280 h:720 pixfmt:yuv420p [scale @ 0x9215100] w:1280 h:720 fmt:yuv420p -> w:320 h:240 fmt:rgb24 flags:0x4 Output #0, gif, to 'output.gif': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 encoder : Lavf53.21.1 Stream #0.0(und): Video: rawvideo, rgb24, 320x240, q=2-31, 200 kb/s, 90k tbn, 10 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Press ctrl-c to stop encoding frame= 101 fps= 32 q=0.0 Lsize= 8686kB time=10.10 bitrate=7045.0kbits/s dup=0 drop=149 video:22725kB audio:0kB global headers:0kB muxing overhead -61.778676% Thanks.

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• ffmpeg conversion problem

  - by user33126

  installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

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• ffmpeg conversion problem

  - by Elamurugan

  installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

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• HP tx2510us shuts down without warning, now won't boot, no BIOS codes, screen doesn't light

  - by Tim S

  Hey all, HP tx2150us worked great for a year and a half, rarely it would shut down instead of sleeping. It started running hot sometimes, I installed Win7, worked great, still running hot. Before Xmas, it gave me evidence of video errors - jagged lines, rows missing. That same night, it started rebooting, then it shut off and wouldn't boot. It gave me a BIOS code, I shut it off to look up the code online. When I turned it on again, it won't boot at all. The LEDs all light up, the fan, HD, and optical all spin up, but the screen never lights up and it doesn't try to boot, nor does it blink any BIOS codes. It just acts like it's sleeping, and won't wake up. I suspected heat problems, so I disassembled it, cleaned the crap out of the fan, and reassembled it, breaking the stereo mic connector. Oh well. When I reassembled it, it booted up into Win7 again, but kept shutting down for no discernible reason. After a dozen or so random reboots like that, it is now back to where it was - turns on but doesn't boot or give BIOS codes. The screen never lights, and everything spins up then idles. Any ideas? I really can't afford to buy a new one and I use(d) it to take ALL my notes, that's why I got a tablet in the first place.

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• No sound Ubuntu 10.10 (have SB Live! card)

  - by Chris Frazier

  I am new to Linux/Ubuntu and I just installed 10.10 on my Dell Desktop PC that was running Windows XP. The PC is about 5-6 years old and the sound card it has is a SoundBlaster Live! card. The Sound Preferences recognizes the card as [SB Live! Value] EMU10k1X. It is currently set to Analog Stereo Duplex. I tried multiple other sound configurations and nothing works. No sound but some clicks come out of the speakers. I ran the system test and during the audio tests the same thing happened, just clicks and pops. I tried to play some music with Rhythmbox and when it tries to play a track it tries for several seconds and then just closes itself, or sometimes just doesn't play. I ran a check for drivers, but the only driver it found to install was for my nVidia video card. It did not show any drivers for the sound card. Does anyone have any idea what I need to do to get the sound to work?

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• Codecs, Premiere Pro & Quicktime: Import or Play Error

  - by Nchpmn

  Original Question I've been using a FS-H200 (not the Pro variant) recorder with a JVC ProHD camera. I have been shooting with the DTE FORMAT to Quicktime (.mov). I copied the files to an external hard drive and am now trying to edit. The files will play back in VLC, as they would be expected to. However they will not import into Adobe Premiere CS5.5, instead giving an error: Unsupported format or damaged file. Quicktime gives the following error when attempting to play the files: Error -2002: a bad public movie atom was found in the movie (Filename) To try and fix this, I have installed the following codec packs: K-Lite Codec Pack 64-bit Full (version 5.9, latest) K-Lite Codec Pack 32-bit Full (version 8.4, latest) MainConcept Codec Suite (Broadcast) v5.1 for Adobe CS5 Reinstalled Quicktime with new download from Apple The same errors and problems still exist. From this I can assume that there is an issue with Quicktime and that is what Premiere is using as an encoder/decoder for the codec. Is there any way to fix this? From looking at the "Codec Information" from VLC: Stream 0 Type: Video Codec: MPEG-1/2 (mpgv) Language: English Resolution: 1280 x 720 Frame Rate: 25 Stream 1 Type: Audio Codec: PCM S16 BE (twos) Language: English Channels: Stereo Sample Rate: 48000 Hz Bits per sample: 16 Other computer specs: Windows 7 Professional 64-bit (SP1) Gigabyte Z68X-UD3-B3 Intel i7-2600K 16GB DDR3 2TB WD 7200RPM SATA 6Gb/s LaCie d2 Quadra 2TB v3 7200RPM (External HDD) NVIDIA GeForce GTX 560 Ti Golden Sample Updates 2012-03-11 @ 2050 AEDT MPEG Steamclip doesn't recognise, play or convert the footage. File open error: unrecognised file type. [Open Anyway] File open error: can't find video or audio tracks. 2012-03-24 @ 1920 AEDT Had to transcode the footage. :(

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• Youtube has no voice but the music continues just fine?

  - by Prix

  PC CONFIG: gigabyte EP45C UD3R with the realtek HD onboard 4gb dual channel Qcore 2.83ghz When i watch to videos on youtube now the voice some times is in static and some times so low that you can hear it while the sound continues just fine... For example if can hear to things like guitar or a train etc but the voice of whoever is speaking is gone or very low or pure static when watching the videos. I know some videos have a really great quality and some are HD 1080p so this was something not expected to happen. I can aswell play videos on my WMP11 just fine i have ccc-p installed also tried k-lite, both on the latest stable avaiable. I havent tried anything else related to flash but something is either wrong with my drivers or youtube. I have installed the latest drivers to make sure they are up-to-date but this didnt help either. What i have tried so far: removed the audio drivers and re-installed remove any codec pack i had and re-installed k-lite, test, didnt worked remove any codec pack i had and re-installed cccp, test, didnt worked checked the control panel sound configurations, tried chaging to phone stereo, to 5.1 which is what my headphone is. checked the realtek manager, tried changing the sound channels from 2CH to 6CH to reflect my headphone, didnt work. rebooted after every change of the above tries. tried chrome, firefox and internet explorer with the same results didnt w

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• What sound cards are there besides Creative's that offer benefits for gaming?

  - by Vilx-

  Many years (6? 7?) ago I bought an Audigy sound card to replace the onboard sound and was astonished at the improvement in games. It was a completely different sound, the whole experience became way more immersive. As the time has passed however the card has become old. The support for the latest Windows versions is declining and newer technologies have definitely been developed. So I was starting to wonder - what newer hardware exists? Sure, there is the Sound Blaster X-Fi, but that's quite expensive and I'm not entirely thrilled by past policies of Creative either (like the whole affair with Daniel_K). But are there any alternatives? EAX is a patent by Creative, so it's doubtful that any other manufacturer has implemented it. And I haven't heard of any competing standards either. To clarify, what I would like is something like a "sound accelerator". A sound card that would offload sound processing from my CPU while at the same time giving astounding effects that would be impractical to do on CPU in the first place. I'm not interested in absurd sampling rates (for the most time I can't tell MP3 and a CD apart) or uncountable channels (I'm using stereo headphones). But I am interested in special effects in games. Are there any alternatives or is Creative a monopoly in this market?

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